Call inot closing, clearing after Calle and Caller drop call

1. Brekeke Product Name and version: Brekeke SIP Server for Brekeke PBX, Version 2.2.6.2

2. Java version: 1.6.0.100

3. OS type and the version: Win2003 Server Standard

4. UA (phone), gateway or other hardware/software involved: IVRCat, Cellphone

5. Select your network pattern from http://www.brekeke-sip.com/bbs/network/ ... terns.html : Call is not closing or clearing/Tearing down after Calle and Caller drops the call. It hangs in status "Talking". I have set the RTP relay to "On" and even "auto" . I have reduced the Configuration > RTP > RTP Session Timeout (ms) = 60000 (1min)

I have Deploy Patterns:
&net.sip.addrecordroute=false
&net.sip.addrecordroute.lr=false
&net.rtp.session.timeout=60000

Why did you set &net.sip.addrecordroute=false and &net.sip.addrecordroute.lr=false ?
It is a cause of the issue. so you don't have to set them.

I added them "&net.sip.addrecordroute=false and &net.sip.addrecordroute.lr=false" while troubleshooting. They have been removed and problem of calls not closing still exists. (PBX restarted after all changes)

The current
Dial PLAN
$request=^INVITE
To=sip:(.*)@

Deplore Pattern
$auth=true
To=sip:%1@sip.sipo.com
$session=com.sample.radius.proxy.RadiusAcct
$continue=false

Are you using any ITSP? Which ITSP is it?

I use "siptraffic"

Hi

Add "$b2bua=true" in the Deplore Patterns..
It may effect..

Thanks JanP,

I have it set as below and i APPLY the rule. however the issue remains

Deplore Pattern
$auth=true
To=sip:%1@sip.sipo.com
$session=com.sample.radius.proxy.RadiusAcct
$b2bua=true
$continue=false

Are you using Brekeke PBX and its ARS?

Why did you set "$auth=true"?
Generally, you need to set "$auth=false" for connecting to an ITSP..

I am usning the "$auth=true" because of my Radius AAA Server. If i set "$auth=false" then unregistered callers will make free calls......lol

Hi all,

If you set the INVITE-AUTHENTICATION in the Config page, you don't have to set "$auth=true" in dialplan.

And I wonder. if a call is coming from the ITSP, can the ITSP send a valid credential to the SIP Server? I mean.. can you make a call from the ITSP to the SIP?

Harold,

You can open a new post for this. However if you set AUTHENTICATION -> INVITE in the Config page "ON it means that all devices must authenticate. However if you set "$auth=true" in the Dial Plan it explicitly means a registered device is only allowed to use that Deployed Pattern. see this post http://www.brekeke-sip.com/bbs/viewtopi ... highlight=

How about this DialPlan ruke?

Deplore Pattern
$auth=true
To=sip:%1@sip.sipo.com
$b2bua=true
$continue=true


I removed "$session=com.sample.radius.proxy.RadiusAcct".
I know you need it for billing. but try it for testing.

I removed "$session=com.sample.radius.proxy.RadiusAcct" and put in the below but the calls did not tear down

Deplore Pattern
$auth=true
To=sip:%1@sip.sipo.com
$b2bua=true
$continue=true

Hi

If you end a call from SIP side, the call is not closed?
If you end a call from ITSP side, the call is not closed?

SIP side close, Cell Phone close, ISP side closes but looking at "Active Calls on BSS i see the session active hence the Billing software continue to bill my client.

Funny thing that happened today. I called my brother in UK from Canada and i hang off my phone and he hangs off his phone, about 5min later he wants to make a new call and as he picked up his phone"OFF Hook" his UK phone started calling me by itself and when i picked the phone he told me the phone called by itself. I belive the session did not end and that was why i was called back.

Am just wondering if Brekeke have a solution for this or they dont look at this forum ???

I connected to a different carrier "B" today to test. This new carrier"B" does not have the current issues of calls not closing that i have with current carrier "A" . I have contacted my carrier "A" to check why their calls are not closing.

What bothers me is that this started happening after i upgraded to latest version 2.2.6.2 Could it be a coincidence ? Maybe something in the new ver code change the way carrier "A" close calls. ALL Hands on DECK !!!

Can you capture SIP packets on the server computer?
And check what kind of SIP packet are exchanged between the SIP Server and the carrier.

If the call is ended from the carrier side, the carrier should send BYE packet to the SIP Server.
If the call is ended from the SIP Server, the carrier should return 200 OK to BYE.

> Am just wondering if Brekeke have a solution for this or they dont look at this forum ???

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